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-rw-r--r--third_party/libwebrtc/modules/audio_processing/agc2/saturation_protector_buffer.cc77
1 files changed, 77 insertions, 0 deletions
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/saturation_protector_buffer.cc b/third_party/libwebrtc/modules/audio_processing/agc2/saturation_protector_buffer.cc
new file mode 100644
index 0000000000..41efdad2c8
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/saturation_protector_buffer.cc
@@ -0,0 +1,77 @@
+/*
+ * Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/agc2/saturation_protector_buffer.h"
+
+#include "rtc_base/checks.h"
+#include "rtc_base/numerics/safe_compare.h"
+
+namespace webrtc {
+
+SaturationProtectorBuffer::SaturationProtectorBuffer() = default;
+
+SaturationProtectorBuffer::~SaturationProtectorBuffer() = default;
+
+bool SaturationProtectorBuffer::operator==(
+ const SaturationProtectorBuffer& b) const {
+ RTC_DCHECK_LE(size_, buffer_.size());
+ RTC_DCHECK_LE(b.size_, b.buffer_.size());
+ if (size_ != b.size_) {
+ return false;
+ }
+ for (int i = 0, i0 = FrontIndex(), i1 = b.FrontIndex(); i < size_;
+ ++i, ++i0, ++i1) {
+ if (buffer_[i0 % buffer_.size()] != b.buffer_[i1 % b.buffer_.size()]) {
+ return false;
+ }
+ }
+ return true;
+}
+
+int SaturationProtectorBuffer::Capacity() const {
+ return buffer_.size();
+}
+
+int SaturationProtectorBuffer::Size() const {
+ return size_;
+}
+
+void SaturationProtectorBuffer::Reset() {
+ next_ = 0;
+ size_ = 0;
+}
+
+void SaturationProtectorBuffer::PushBack(float v) {
+ RTC_DCHECK_GE(next_, 0);
+ RTC_DCHECK_GE(size_, 0);
+ RTC_DCHECK_LT(next_, buffer_.size());
+ RTC_DCHECK_LE(size_, buffer_.size());
+ buffer_[next_++] = v;
+ if (rtc::SafeEq(next_, buffer_.size())) {
+ next_ = 0;
+ }
+ if (rtc::SafeLt(size_, buffer_.size())) {
+ size_++;
+ }
+}
+
+absl::optional<float> SaturationProtectorBuffer::Front() const {
+ if (size_ == 0) {
+ return absl::nullopt;
+ }
+ RTC_DCHECK_LT(FrontIndex(), buffer_.size());
+ return buffer_[FrontIndex()];
+}
+
+int SaturationProtectorBuffer::FrontIndex() const {
+ return rtc::SafeEq(size_, buffer_.size()) ? next_ : 0;
+}
+
+} // namespace webrtc